Hi there,
I am wondering is there a preset command to saydecimal number? Currently if
you put comand in dialplan as SayNumber(1234) it will repeat to you. But how
about if the number is decimal like 12.34. Is there any command?
Thanks
--
Regards,
Sharon Lim
*Good memories are to be folded
:
In article [EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Thanks, will do more research on that part. By the way, Im trying to do
IVR
where caller enter the pin the retrieve some information out of the MS
SQL.
I am wondering, what is the constraints or how to go about
Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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of the DB through IVR. Any guidance?
Thanks. On 11/14/06, Vicky [EMAIL PROTECTED] wrote:
oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield
[EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED]
,Sharon Lim
[EMAIL PROTECTED] wrote: Hi there, I am looking around
-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket
focus?
Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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:
Sharon,
pbxware.bicomsystems.com
U: [EMAIL PROTECTED]
P: pbxware
All standard.
Steve
steve {at] bicomsystems [dot} com
- Original Message -
From:
Sharon Lim
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, September 28, 2006 9:52
AM
Subject
40 23114549F:+91 40 40208727
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-- Regards, Sharon Lim *Good memories
eer
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back poc
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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have better luck searching there.
bp
On 9/11/06, Steve Totaro [EMAIL PROTECTED]
wrote:
Sharon Lim wrote: Hi all, I have tried to install freepbx and a2billing application. Now see
both application is not integrated special on cdr part. Any idea how to integrated it?Confuse! -- Regards, Sharon
by Easynews.com
--asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse!-- Regards,
Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket
http://www.trixbox.org/modules/smartsection/item.php?itemid=4 On 8/28/06, Rizwan Hisham
[EMAIL PROTECTED] wrote:
hi guys,
i need to install the .tar.gz version of trixbox. i cant find any help files for installation in it and also there is no help for it on the website. can anybody please help?
I am not sure whether username can be xyz.abc cause normally is single words. try to change it. On 8/14/06, Crazy Boy
[EMAIL PROTECTED] wrote:Hi,Thank you for your response. As you said, I executed the command sip show registry. But, its not showing anything. Teliax people are also telling that
is there something wrong with ur syntax at exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) exten = _1XX,1,DIAL(SIP/teliax,${EXTEN},30,tr)
On 8/14/06, Crazy Boy
[EMAIL PROTECTED] wrote:
Hi, My user name is : rudy.pandya Thank you.Sharon Lim
[EMAIL PROTECTED] wrote: I am not
http://www.itelbilling.com/ try this!
On 8/12/06, Wasif [EMAIL PROTECTED] wrote:
Hello,Does anyone knowabout open source wholesale billing for Asterisk?Thanks___--Bandwidth and Colocation provided by
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If i am not mistaken, you need to have another IAX user tables to store all the iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!On 8/9/06, Shaun [EMAIL PROTECTED] wrote:
I'm attempting to setup asterisk running real-time with mysql.Right now Ican
] wrote:
IAX is being read from the flat config like it
normally is. I can verify this because asterisk registers with my
provider.
-- ~Shaun
Sharon Lim [EMAIL PROTECTED] wrote
in message news:[EMAIL PROTECTED]
...If
i am not mistaken, you need to have another IAX user tables to store
Did you leave any message in your voicemail? Cause by default if 10 seconds silence then it will end the recording. On 8/8/06, ismir saljic
[EMAIL PROTECTED] wrote:Hi , I have the problem with voicemail message
duration.Every message is only 10 seconds long.
just wanted to let you know you
I am trying to do a simple agi connection to db with the guidance from http://www.voip-info.org/wiki/view/Asterisk+AGI+php Item 13 with ani.agi file, db and
extensions.conf13. another sample, ANI
Scenario - did callers call the Asterisks box and land on
the context did, Asterisks answers the
I have an old pbx and I want to pass callerid frm the old pbx to asterisk as a voicemail server. My old pbx have sent the callerid but i am not sure how to make it into dialpattern cause if I have 1000 callerid then i have to enter 1000 enter into
extensions.conf. I m using tdm400p where i pull
Hello, Don and steven, Something like this should drop you to the voicemail box of the caller ID.
exten = s,1,Voicemail(${CALLERIDNUM})Don PobanzThe previous settings cant get any calleridnum. But this can
exten = XXX,1, Voicemail(${EXTEN}) may work, but you have to have it as _XXXThe underscore
have 2 company and want to have same sip account context, how do i differentiate with it? Thanks in advance. On 7/17/06, El Flynn
[EMAIL PROTECTED] wrote:
Sharon Lim wrote: Hi there, I would like to ask, is it possible to group sip user? Means group A with sip user 100,200 and group B with sip
Perhaps you can check on the dtmf code. On 7/14/06, Robert La Ferla [EMAIL PROTECTED] wrote:
When I dial out, I can't hear any ringing.I am using the latest SVNcode (SVN-branch-1.2-r37458M
).Is this a problem with Asterisk? Orwith my VOIP
Hi there, I would like to ask, is it possible to group sip user? Means group A with sip user 100,200 and group B with sip user 100,200? thanks in advance.
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and will ring which user phone. Cause if we have 2 jenny then if in extensions.conf exten=200,1,Dial (SIP/jenny) which account will ring. That's my problem.
Thanks so much for the feedback.On 7/6/06, Olle E Johansson [EMAIL PROTECTED] wrote:
6 jul 2006 kl. 04.09 skrev Sharon Lim: hi, Is it possible
hi, Is it possible to have same sip context but refering to different username and context? Is it a must to have username(test) and the sip context [test] the same? what the different with username sip context?
can username variable be alphanumeric, like using email address as username?default
anyone have information on how the call back features work with asterisk? I means the dial plan or what so ever. thanks
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Hi John, Your first question, I am not sure why but for this part i can explain abit
Also, on a side note, I have a context called [home] which each SIPPhone is associated with.Do I need to specify each extension inthere?SIP user can register as name as well . Doesnt means to have number.
Did you install the sound packages such as mpg123-0.59r-1.i386.rpm ? Can download from http://rpm.pbone.net/index.php3/stat/4/idpl/516450/com/mpg123-0.59r-1.i386.rpm.html
good luck!On 6/16/06, kharris [EMAIL PROTECTED] wrote:
Can anyone point me in the direction for resources for
I had the same problem. I change some variable in zapata.conf such as in [defaults] context : 1. busydetect=yes2. busycount=43. hanguponpolarityswitch=yes ; some said need this variables4. rxgain=
1.0 5. txgain=1.0not sure which one effect it...but tried it...On 6/16/06, Steven Ringwald [EMAIL
yes. you can use xten http://www.xten.net/index.php?menu=download. free to download.On 6/15/06, Asterisk guy
[EMAIL PROTECTED] wrote:
are there any open source sip softphone (Window OS version )?___--Bandwidth and Colocation provided by Easynews.com
://www.snapanumber.com
On 6/9/06, Sharon Lim [EMAIL PROTECTED] wrote: Hi colin, I am doing on php. But i would glad that you can share the codes as i will
explore it. Thanks. On 6/9/06, Colin Anderson [EMAIL PROTECTED] wrote:
I have, using Active Server Pages + Flash. See: http
each ip phone need to register to be able to call the conference. Firstly, you need to create user with username and password under sip.conf, Then you need to create conference room which is meetme in meetme.conf
then you need to create extension to point to the conference room in
Hi there,anyone in the community has manage to configure click to call features? Care to share.I have tried on this manual , seem got some software error like
http://www.voip-info.org/wiki/view/Asterisk+click+to+callSoftware error:
Unable to determine call statusMessage: Originate with 'Exten'
the .asp and .fla somewhere
if someone is interested in it.
-Original Message-From: Sharon Lim
[mailto:[EMAIL PROTECTED]]Sent: Friday, June 09, 2006 6:37
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] click to call features
Hi there, Try use Web-MeetMe http://www.voip-info.org/wiki/view/MeetMe-Web-Control . I have tried to install but havent had time to configure. The latest version has a Conference schedulling..maybe this will helpwhen you get it working maybe can email me the configuration
Hi there, SIP more commanly used and it is a openstandard. Meanwhile IAX2 is a protocol on asterisk. I dont think it will effect the cpu resources cause they are bid with the same codecs like G711 and etc..so if you used SIP or IAX2 it also refer to the same codecs...so dont think it will take a
Hmm...any idea where to define the context of a conference? Cause from my understanding, [rooms] context is a default. ThanksOn 5/18/06, Gavin Henry
[EMAIL PROTECTED] wrote:
quote who=Sharon Lim hi there, i am wondering can meetme.conf able to support diffferent context. Cause currently, it has
hi there, i am wondering can meetme.conf able to support diffferent context. Cause currently, it has [rooms] context. ]is it possible to have same conference number with different context?thanks
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Are you looking for an web interface that write to asterisk config files? if yes, you can look at freepbx.org . On 5/11/06,
Kerry Garrison [EMAIL PROTECTED] wrote:
You could install any number of interfaces but it does not
come with one.
Kerry
GarrisonDirector of Technical
I am sorry cause i post this questions is not related to your problem, but i am having problem detecting my TDM400P which is a TDM400P problem. I manage to installed the card with compiling with zaptel and it got 2FXS and 2FXO.
I am having problem while reboot or restart the system. Kudzu seem to
Hi there, I am new to asterisk, I am trying to write a dialplan with 2 process running concurrently. Current dialplan only able to execute process with priority example exten =100,1,Answer()exten =100,2,Musiconhold()
exten =100,3,Hangupis it possible to have process musiconhold/background and dial
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